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Jonathan

2004-Aug-10 02:12 UTC

[Asterisk-Users] CAPI call transfer

Hi,

I am having trouble configuring CAPI so that call transfers work.
I make a SIP call to asterisk which goes out on ISDn via CAPI.  Then
I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.

The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards.
Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack
    -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0x9ee
    -- Called 01824708169:01824708752
    -- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8
    -- CAPI[contr1/01824708169]/11 is ringing
    -- CAPI[contr1/01824708169]/11 answered
SIP/192.168.1.162-08186af8
  == Spawn extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'
    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:h") in new stack
    -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0xa5c
    -- Called 01824708169:h
    -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack
Aug 10 09:55:29 NOTICE[442386]: chan_capi.c:1172 capi_request: didn't
find capi device with outgoing msn = 01824708169. you should check
your config!
Aug 10 09:55:29 NOTICE[442386]: app_dial.c:706 dial_exec: Unable to
create channel of type 'CAPI'
  == Everyone is busy/congested at this time
    -- Executing Congestion("CAPI[contr1/01824708169]/11",
"") in new
stack
    -- CAPI Hangingup
       > sent DISCONNECT_REQ PLCI=0x201
    -- removed pipe for PLCI = 0x201
  == Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf
[general]
nationalprefix=0
internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=*
softdtmf=1

mode=immediate
isdnmode=ptp
msn=01824708,01824708169
controller=1
devices=2

Roland Zagler

2004-Aug-10 02:18 UTC

head link

[Asterisk-Users] CAPI call transfer

Try specifying your number you want to dial with "b" in front of, e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards,
roland 


Roland Zagler
mailto:laureen@laureen.at
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] CAPI call transfer

Hi,

I am having trouble configuring CAPI so that call transfers work.
I make a SIP call to asterisk which goes out on ISDn via CAPI.  Then I
try to do a transfer from the SIP phone which doesn't work and results
in the call being disconnected.

The error message given by asterisk is that it chan_capi can't find an
entry for the outgoing msn for the transfer however the outgoing msn is
the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards.
Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack
    -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0x9ee
    -- Called 01824708169:01824708752
    -- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8
    -- CAPI[contr1/01824708169]/11 is ringing
    -- CAPI[contr1/01824708169]/11 answered
SIP/192.168.1.162-08186af8
  == Spawn extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'
    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:h") in new stack
    -- creating pipe for PLCI=-1
       > sent CONNECT_REQ MN =0xa5c
    -- Called 01824708169:h
    -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack
Aug 10 09:55:29 NOTICE[442386]: chan_capi.c:1172 capi_request: didn't
find capi device with outgoing msn = 01824708169. you should check your
config!
Aug 10 09:55:29 NOTICE[442386]: app_dial.c:706 dial_exec: Unable to
create channel of type 'CAPI'
  == Everyone is busy/congested at this time
    -- Executing Congestion("CAPI[contr1/01824708169]/11",
"") in new
stack
    -- CAPI Hangingup
       > sent DISCONNECT_REQ PLCI=0x201
    -- removed pipe for PLCI = 0x201
  == Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf
[general]
nationalprefix=0
internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=*
softdtmf=1

mode=immediate
isdnmode=ptp
msn=01824708,01824708169
controller=1
devices=2

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Jonathan

2004-Aug-10 02:36 UTC

head link

[Asterisk-Users] CAPI call transfer

Thanks for your reply Roland, unfortunately adding the 'b' didn't
make
 any diference.

Regards.
Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at>
==> Date: Tue, 10 Aug 2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of,
e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I
make a SIP call to asterisk which goes out on ISDn via CAPI.  Then I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.

The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing
msn = 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

_______________________________________________ Asterisk-Users
mailing
list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To
UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 _______________________________________________ Asterisk-Users
mailing list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To
UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Roland Zagler

2004-Aug-10 02:48 UTC

head link

[Asterisk-Users] CAPI call transfer

Are you using kernel 2.6.x ?


Roland Zagler
mailto:laureen@laureen.at

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM
To: asterisk-users@lists.digium.com
Subject: Re: RE: [Asterisk-Users] CAPI call transfer


Thanks for your reply Roland, unfortunately adding the 'b' didn't
make
any diference.

Regards.
Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of, e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I make
a SIP call to asterisk which goes out on ISDn via CAPI.  Then I try to
do a transfer from the SIP phone which doesn't work and results in the
call being disconnected.

The error message given by asterisk is that it chan_capi can't find an
entry for the outgoing msn for the transfer however the outgoing msn is
the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn
= 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

_______________________________________________ Asterisk-Users mailing
list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 _______________________________________________ Asterisk-Users mailing
list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Jonathan

2004-Aug-10 03:15 UTC

head link

[Asterisk-Users] CAPI call transfer

No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at>
==> Date: Tue, 10 Aug 2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users]
CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of,
e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I
make a SIP call to asterisk which goes out on ISDn via CAPI.  Then I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.

The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing
msn = 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

_______________________________________________ Asterisk-Users
mailing
list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To
UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 _______________________________________________ Asterisk-Users
mailing list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To
UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________ Asterisk-Users
mailing
list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To
UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 _______________________________________________ Asterisk-Users
mailing list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To
UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Roland Zagler

2004-Aug-10 03:21 UTC

head link

[Asterisk-Users] CAPI call transfer

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.htm
l


Roland Zagler
mailto:laureen@laureen.at

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM
To: asterisk-users@lists.digium.com
Subject: Re: RE: RE: [Asterisk-Users] CAPI call transfer


No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make
any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of, e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I make
a SIP call to asterisk which goes out on ISDn via CAPI.  Then I try to
do a transfer from the SIP phone which doesn't work and results in the
call being disconnected.

The error message given by asterisk is that it chan_capi can't find an
entry for the outgoing msn for the transfer however the outgoing msn is
the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn
= 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

_______________________________________________ Asterisk-Users mailing
list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

 _______________________________________________ Asterisk-Users mailing
list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

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list Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE
or update options visit:
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 _______________________________________________ Asterisk-Users mailing
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Jonathan

2004-Aug-10 04:29 UTC

head link

[Asterisk-Users] CAPI call transfer

Hi Roland,

Nothing on that message helps me unfortunately.  I can make calls
from
SIP to ISDN I just can't get call transfer to work. 


Regards, Jonathan


-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at>
==> Date: Tue, 10 Aug 2004 12:21:29  0200

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.
htm l

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: [Asterisk-Users]
CAPI call transfer

 No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users]
CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of,
e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I
make a SIP call to asterisk which goes out on ISDn via CAPI.  Then I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.

The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing
msn = 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

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Roland Zagler

2004-Aug-10 04:35 UTC

head link

[Asterisk-Users] CAPI call transfer

Can you post your extensions.conf, maybe i can find something! 


Roland Zagler
mailto:laureen@laureen.at
mobile:4369910713694
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 1:30 PM
To: asterisk-users@lists.digium.com
Subject: Re: RE: RE: RE: [Asterisk-Users] CAPI call transfer

Hi Roland,

Nothing on that message helps me unfortunately.  I can make calls from
SIP to ISDN I just can't get call transfer to work. 


Regards, Jonathan


-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 12:21:29  0200

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.
htm l

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: [Asterisk-Users]
CAPI call transfer

 No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make
any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of, e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I make
a SIP call to asterisk which goes out on ISDn via CAPI.  Then I try to
do a transfer from the SIP phone which doesn't work and results in the
call being disconnected.

The error message given by asterisk is that it chan_capi can't find an
entry for the outgoing msn for the transfer however the outgoing msn is
the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn
= 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

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Jonathan

2004-Aug-10 05:02 UTC

head link

[Asterisk-Users] CAPI call transfer

My extensions.conf is:
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface
for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest                                   ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2                                    ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip
(usually 1 or 0)
TRUNK=capi
;TRUNK=IAX2/user:pass@provider

[SIP]
exten => _.,1,Dial,CAPI/01824708${CALLERIDNUM}:b${EXTEN}
exten => _.,2,congestion
exten => _.,3,hangup

My sip.conf is:
[general]
context=default 
autocreatepeer=yes
localnet=192.168.1.162
port=5062    
bindaddr=0.0.0.0
rtptimeout=60                             
rtpholdtimeout=300
useragent=PBX Gateway

[sip_proxy]
context=SIP
type=peer
Host=192.168.1.162


Thanks and best regards.
Jonathan 

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at>
==> Date: Tue, 10 Aug 2004 13:35:32  0200

Can you post your extensions.conf, maybe i can find something! 

 Roland Zagler mailto:laureen@laureen.at mobile:4369910713694
-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 1:30 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: RE:
[Asterisk-Users] CAPI call transfer

Hi Roland,

Nothing on that message helps me unfortunately.  I can make calls
from
SIP to ISDN I just can't get call transfer to work. 

 Regards, Jonathan

 -------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 12:21:29  0200

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.
htm l

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: [Asterisk-Users]
CAPI call transfer

 No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users]
CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of,
e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I
make a SIP call to asterisk which goes out on ISDn via CAPI.  Then I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.

The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing
msn = 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

_______________________________________________ Asterisk-Users
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list Asterisk-Users@lists.digium.com
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Roland Zagler

2004-Aug-10 05:13 UTC

head link

[Asterisk-Users] CAPI call transfer

Have you tried removing "${CALLERIDNUM}" from your 1st line in context
[SIP] in extensions.conf? 
Is your ISDN Line configured to transfer the Extensions to you
(Provider-dependent)?
And try to put "Answer" before calling to CAPI!

I do it like this:

[MyContext1]
exten => _.,1,Answer
exten => _.,2,Dial,CAPI/50:b${EXTEN},60
exten => _.,100,Hangup


Roland Zagler
mailto:laureen@laureen.at
mobile:4369910713694
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 2:03 PM
To: asterisk-users@lists.digium.com
Subject: Re: RE: RE: RE: RE: [Asterisk-Users] CAPI call transfer


My extensions.conf is:
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface
for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest                                   ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2                                    ; Trunk interface
TRUNKMSD=1                                      ; MSD digits to strip
(usually 1 or 0)
TRUNK=capi
;TRUNK=IAX2/user:pass@provider

[SIP]
exten => _.,1,Dial,CAPI/01824708${CALLERIDNUM}:b${EXTEN}
exten => _.,2,congestion
exten => _.,3,hangup

My sip.conf is:
[general]
context=default
autocreatepeer=yes
localnet=192.168.1.162
port=5062    
bindaddr=0.0.0.0
rtptimeout=60                             
rtpholdtimeout=300
useragent=PBX Gateway

[sip_proxy]
context=SIP
type=peer
Host=192.168.1.162


Thanks and best regards.
Jonathan 

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 13:35:32  0200

Can you post your extensions.conf, maybe i can find something! 

 Roland Zagler mailto:laureen@laureen.at mobile:4369910713694
-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 1:30 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: RE:
[Asterisk-Users] CAPI call transfer

Hi Roland,

Nothing on that message helps me unfortunately.  I can make calls from
SIP to ISDN I just can't get call transfer to work. 

 Regards, Jonathan

 -------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 12:21:29  0200

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.
htm l

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: [Asterisk-Users]
CAPI call transfer

 No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make
any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of, e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I make
a SIP call to asterisk which goes out on ISDn via CAPI.  Then I try to
do a transfer from the SIP phone which doesn't work and results in the
call being disconnected.

The error message given by asterisk is that it chan_capi can't find an
entry for the outgoing msn for the transfer however the outgoing msn is
the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn
= 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

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Jonathan

2004-Aug-10 05:37 UTC

head link

[Asterisk-Users] CAPI call transfer

Hi Roland,

Still no difference.  The call works fine but the transfer fails with
the same error message as before: 
 -- Executing Dial("CAPI[contr1/01824708169]/0",
"CAPI/01824708169:b170") in new stack
Aug 10 13:34:34 NOTICE[294930]: chan_capi.c:1172 capi_request: didn't
find capi device with outgoing msn = 01824708169. you should check
your config!

I have "${CALLERIDNUM}" so my SIP phones are mapped to DDI's. This
avoids having to have an msn entry for every phone with a DDI.

Thanks
Jonathan 


-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at>
==> Date: Tue, 10 Aug 2004 14:13:04  0200

Have you tried removing "${CALLERIDNUM}" from your 1st line in
context
[SIP] in extensions.conf? Is your ISDN Line configured to transfer
the
Extensions to you (Provider-dependent)? And try to put "Answer"
before
calling to CAPI!

I do it like this:

[MyContext1] exten => _.,1,Answer exten =>
_.,2,Dial,CAPI/50:b${EXTEN},60 exten => _.,100,Hangup

 Roland Zagler mailto:laureen@laureen.at mobile:4369910713694
-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 2:03 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: RE: RE:
[Asterisk-Users] CAPI call transfer

 My extensions.conf is: [general] static=yes writeprotect=no

[globals] CONSOLE=Console/dsp                             ; Console
interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest
                                  ; IAXtel username/password
;IAXINFO=myuser:mypass TRUNK=Zap/g2                                  
 ; Trunk interface TRUNKMSD=1                                      ;
MSD digits to strip (usually 1 or 0) TRUNK=capi
;TRUNK=IAX2/user:pass@provider

[SIP] exten => _.,1,Dial,CAPI/01824708${CALLERIDNUM}:b${EXTEN} exten
=> _.,2,congestion exten => _.,3,hangup

My sip.conf is: [general] context=default autocreatepeer=yes
localnet=192.168.1.162 port=5062 bindaddr=0.0.0.0 rtptimeout=60
rtpholdtimeout=300 useragent=PBX Gateway

[sip_proxy] context=SIP type=peer Host=192.168.1.162

 Thanks and best regards. Jonathan 

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 13:35:32  0200

Can you post your extensions.conf, maybe i can find something! 

 Roland Zagler mailto:laureen@laureen.at mobile:4369910713694
-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 1:30 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: RE:
[Asterisk-Users] CAPI call transfer

Hi Roland,

Nothing on that message helps me unfortunately.  I can make calls
from
SIP to ISDN I just can't get call transfer to work. 

 Regards, Jonathan

 -------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 12:21:29  0200

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.
htm l

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: [Asterisk-Users]
CAPI call transfer

 No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users]
CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of,
e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I
make a SIP call to asterisk which goes out on ISDn via CAPI.  Then I
try to do a transfer from the SIP phone which doesn't work and
results
in the call being disconnected.

The error message given by asterisk is that it chan_capi can't find
an
entry for the outgoing msn for the transfer however the outgoing msn
is the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing
msn = 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

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Roland Zagler

2004-Aug-10 11:09 UTC

head link

[Asterisk-Users] CAPI call transfer

You could try to specify incomingmsn *NOT* to "*" and outgoingmsn in
your capi.conf


Roland Zagler
mailto:laureen@laureen.at
mobile:4369910713694
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 2:38 PM
To: asterisk-users@lists.digium.com
Subject: Re: RE: RE: RE: RE: RE: [Asterisk-Users] CAPI call transfer


Hi Roland,

Still no difference.  The call works fine but the transfer fails with
the same error message as before: 
 -- Executing Dial("CAPI[contr1/01824708169]/0",
"CAPI/01824708169:b170") in new stack
Aug 10 13:34:34 NOTICE[294930]: chan_capi.c:1172 capi_request: didn't
find capi device with outgoing msn = 01824708169. you should check your
config!

I have "${CALLERIDNUM}" so my SIP phones are mapped to DDI's. This
avoids having to have an msn entry for every phone with a DDI.

Thanks
Jonathan 


-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 14:13:04  0200

Have you tried removing "${CALLERIDNUM}" from your 1st line in context
[SIP] in extensions.conf? Is your ISDN Line configured to transfer the
Extensions to you (Provider-dependent)? And try to put "Answer"
before
calling to CAPI!

I do it like this:

[MyContext1] exten => _.,1,Answer exten =>
_.,2,Dial,CAPI/50:b${EXTEN},60 exten => _.,100,Hangup

 Roland Zagler mailto:laureen@laureen.at mobile:4369910713694
-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 2:03 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: RE: RE:
[Asterisk-Users] CAPI call transfer

 My extensions.conf is: [general] static=yes writeprotect=no

[globals] CONSOLE=Console/dsp                             ; Console
interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest
                                  ; IAXtel username/password
;IAXINFO=myuser:mypass TRUNK=Zap/g2                                  
 ; Trunk interface TRUNKMSD=1                                      ;
MSD digits to strip (usually 1 or 0) TRUNK=capi
;TRUNK=IAX2/user:pass@provider

[SIP] exten => _.,1,Dial,CAPI/01824708${CALLERIDNUM}:b${EXTEN} exten =>
_.,2,congestion exten => _.,3,hangup

My sip.conf is: [general] context=default autocreatepeer=yes
localnet=192.168.1.162 port=5062 bindaddr=0.0.0.0 rtptimeout=60
rtpholdtimeout=300 useragent=PBX Gateway

[sip_proxy] context=SIP type=peer Host=192.168.1.162

 Thanks and best regards. Jonathan 

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 13:35:32  0200

Can you post your extensions.conf, maybe i can find something! 

 Roland Zagler mailto:laureen@laureen.at mobile:4369910713694
-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 1:30 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: RE:
[Asterisk-Users] CAPI call transfer

Hi Roland,

Nothing on that message helps me unfortunately.  I can make calls from
SIP to ISDN I just can't get call transfer to work. 

 Regards, Jonathan

 -------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 12:21:29  0200

Here's the post i used to get this thing going, maybe it helps:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41324.
htm l

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 12:16 PM To:
asterisk-users@lists.digium.com Subject: Re: RE: RE: [Asterisk-Users]
CAPI call transfer

 No I'm on kernal 2.4.22 Fedora core 1.

Thanks, Jonathan

-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:48:17  0200

Are you using kernel 2.6.x ?

 Roland Zagler mailto:laureen@laureen.at

-----Original Message----- From:
asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:36 AM To:
asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] CAPI
call transfer

 Thanks for your reply Roland, unfortunately adding the 'b' didn't
make
any diference.

Regards. Jonathan



-------- Original Message --------

==> From: "Roland Zagler" <laureen@laureen.at> ==> Date:
Tue, 10 Aug
2004 11:18:32  0200

Try specifying your number you want to dial with "b" in front of, e.g.
"Dial(CAPI/01824708169:b01824708752,60)" in your extensions.conf!

Regards, roland 

 Roland Zagler mailto:laureen@laureen.at -----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan
Sent: Tuesday, August 10, 2004 11:12 AM To:
asterisk-users@lists.digium.com Subject: [Asterisk-Users] CAPI call
transfer

Hi,

I am having trouble configuring CAPI so that call transfers work. I make
a SIP call to asterisk which goes out on ISDn via CAPI.  Then I try to
do a transfer from the SIP phone which doesn't work and results in the
call being disconnected.

The error message given by asterisk is that it chan_capi can't find an
entry for the outgoing msn for the transfer however the outgoing msn is
the same as that used to make the original call.

Has anyone got any ideas please?

The asterisk trace and my capi.conf are below:

Thank you. Best regards. Jonathan 

    -- Executing Dial("SIP/192.168.1.162-08186af8",
"CAPI/01824708169:01824708752") in new stack -- creating pipe for
PLCI=-1 > sent CONNECT_REQ MN =0x9ee -- Called
01824708169:01824708752
-- CAPI[contr1/01824708169]/11 is making progress passing it to
SIP/192.168.1.162-08186af8 -- CAPI[contr1/01824708169]/11 is ringing
-- CAPI[contr1/01824708169]/11 answered SIP/192.168.1.162-08186af8 =Spawn
extension (SIP, 01824708752, 1) exited non-zero on
'SIP/192.168.1.162-08186af8' -- Executing
Dial("SIP/192.168.1.162-08186af8", "CAPI/01824708169:h") in
new stack
-- creating pipe for PLCI=-1 > sent CONNECT_REQ MN =0xa5c -- Called
01824708169:h -- Executing Dial("CAPI[contr1/01824708169]/11",
"CAPI/01824708169:170") in new stack Aug 10 09:55:29 NOTICE[442386]:
chan_capi.c:1172 capi_request: didn't find capi device with outgoing msn
= 01824708169. you should check your config! Aug 10 09:55:29
NOTICE[442386]: app_dial.c:706 dial_exec: Unable to create channel of
type 'CAPI' == Everyone is busy/congested at this time -- Executing
Congestion("CAPI[contr1/01824708169]/11", "") in new stack
-- CAPI
Hangingup > sent DISCONNECT_REQ PLCI=0x201 -- removed pipe for PLCI 0x201 ==
Spawn extension (SIP, h, 1) exited non-zero on
'SIP/192.168.1.162-08186af8'

capi.conf [general] nationalprefix=0 internationalprefix=44
rxgain=0.8
txgain=0.8

[interfaces]

incomingmsn=* softdtmf=1

mode=immediate isdnmode=ptp msn=01824708,01824708169 controller=1
devices=2

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